PacketGen™ Brochure

PacketGen™
(SIP Bulk Call Generator)
SIP Bulk Call Generation with
or without RTP
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RTP Traffic Generation
(Voice, Fax, Digits, Tones)
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Test SIP Phones, Proxy
Servers, Registrars, PSTN, &
Media Gateway
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Overview
PacketGen™ is a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP
generation) for stress testing and precise analysis of the VoIP network equipment. PacketGen™ is
based on a distributed architecture, wherein SIP and RTP software cores can be modularly
stacked in one or many PCs to create a scalable high capacity test system. PacketGen™ can be
used to test basic functionality and verify proper protocol implementation in SIP based
Scalable Distributed
equipment such as SIP phones and Network servers, Proxy Servers, Registrar Servers, and PSTN
Architecture allows Higher Call
and Media Gateways.
Density
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GL’s PacketGen breaks ground with high density performance:
PacketGen™ on an i7 PC can support 2000 simultaneous calls with both SIP and RTP generation.
This performance number is associated with using the G.711 codec, while other codecs may
Supports Almost All Industry
provide higher call densities. PacketGen’s distributed architecture achieves higher call density by
Standard Codec
interconnection multiple computer systems with SIP and RTP software cores on each.
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For more details, visit http://www.gl.com/packetgen.html.
SIP functionality –
Registration, Call Hold, Call
Forwarding, Authentication
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Scripting for RTP Traffic
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Load Testing with High Call
Rates and Media Streams
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Manual and Bulk Call
Generation
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Main Features
Capacity
Call
Generation
Traffic
Handling
Supported
Codecs
Remote
Access
Reports
Remote Access Capability
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Additional
Utilities
 Distributed architecture for SIP and RTP systems provides high call rates and
media streams. Also, makes it scalable, i.e., easy to add additional load
generation capacity.
 PacketGen™ breaks ground with high density performance by generating 2000
simultaneous calls on an i7 PC. Higher density is also achievable using multiple
systems.
 RFC 3261 compliant, RFC 2833 digit generation/detection.
 Generates both SIP signaling and RTP traffic.
 Send / Record voice files on any (or all) RTP sessions. Recorded voice files can
be sent to VQT software for voice quality analysis (requires additional license)
 All Codec supported including - G.711, G.711 App II with VAD, G.729, G.726,
G.726 with VAD, GSM, AMR NB and WB, EVRC, SMV, iLBC, SPEEX NB and WB,
G722, and G722.1. Visit Voice Codecs webpage for more comprehensive
information.
 Remote access capability using GUI or command line interface or through
Remote Desktop
 Provides statistics, events and call records
 GL’s Audio File Conversion Utility (Audio FCU) is used in conjunction with GL’s
packet products to support send and record voice file features.
 GL’s Audio Stream Utility is used to playback the selected calls from a remote
computer to the speaker on the local client PC.
RTP Impairments Generation
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Statistics, Events and Call
Records
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Document Number: PKS100-14.7.25-01
818 West Diamond Avenue - Third Floor, Gaithersburg, MD 20878, U.S.A
(Web) http://www.gl.com/ - (V) +1-301-670-4784 (F) +1-301-670-9187 - (E-Mail) [email protected]
Page 2
SIP Setup and Configuration
User Agent (UA) Configuration Parameters
Various parameters can be configured. They are grouped into 4 tabs:
The SIP Setup screen controls the foundation of the desired test
environment. The user has the flexibility to configure multiple SIP  SIP Parameters – Allows user to set certain SIP/SDP headers for
outgoing messages. User can set UA Name, Host Name, Port,
and RTP instances on a local system and/or remote systems. Each
Sip Server Address, NAT option, and multiple contact entries for
SIP and RTP instance provides additional call density capabilities,
each UA.
thus allowing a true distributed architecture. In addition, true RTP
Load sharing is provided within PacketGen™.
 Media Parameters – Allows user to set the RTP media
parameters (RTP) for the User Agent. These parameters indicate
the media capabilities of the User Agent. These are used to
negotiate media characteristics of the call during call
establishment.
 Extra Headers – Configures extra SIP/SDP headers to be used
for each User Agent. These headers are non-critical headers,
and will be included as is, in the appropriate messages sent for
this User Agent.
 UAC Authentication – Configures the user authentication
information required for UAC simulation.
Figure: SIP Setup Configuration
Manual and Bulk Call Generation
PacketGen™ supports both Manual and Bulk Call Generation, with
complete flexibility on each individual call session such as quick
configuration utility, current status of each configured session,
traffic generation and QOS measurements, call processing options
including hold and call transfer.
Figure: Manual and Bulk Call Generation
Figure: UA Config - SIP Parameters
SIP Registration
PacketGen™ provides facility to register a single or a bunch of User
Agents simultaneously. Each Registration gives flexible
configuration options like Registrar server address, Address of
Record, Expiry time etc. Also, each Registration can be configured
for automatic registration refresh, after the existing registration
expires. A quick configuration utility helps to configure hundreds
of registrations easily.
Figure: UA Registration Configuration
818 West Diamond Avenue - Third Floor, Gaithersburg, MD 20878, U.S.A
(Web) http://www.gl.com/ - (V) +1-301-670-4784 (F) +1-301-670-9187 - (E-Mail) [email protected]
Document Number: PKS100-14.7.25-01
Page 3
Auto Signaling Action
This feature provides a quick and easy method to configure
signaling actions, to be performed automatically as soon as the
call session is established. Configuration is based on call sessions,
thus each call may be configured for unique activities.
Signaling options include Call Transfer, Call Reject (User-Defined
Error), Hold and Re-Direct.
RTP Impairment Generation
PacketGen™ allows user to configure various impairments on
outgoing RTP streams. These categories of impairments can be
generated.
 Latency
 Packet Loss
 Packet Effects
Figure: Auto Traffic and Signaling Actions
Reliable Provisional Responses (RPR)
The ability to send "reliable provisional responses" and start early
media actions. We have two options in Reliable Provisional
Responses Viz: Required and Supported. Below diagram shows a
SIP call flow with RPR's and early media.
Figure: RTP Impairment Generation
Figure: Reliable Provisional Responses (RPR)
Traffic Generation (RTP)
Once the call is established, PacketGen™ can generate and handle
multitude of traffic, either manually or automatically. It facilitates
the mechanism to test various network conditions and responses.
Traffic options include Send Actions, Loop Back, Receive Actions,
and Power Measurement. Auto-Action feature provides a quick
and easy method to configure signaling as well as traffic actions,
once the call session is established.
Advanced traffic options like codec parameters, ptime and Rx
jitter buffer control are provided.
Figure: RTP Traffic Generation
Statistics & Events
PacketGen™ provides detailed statistics for each User Agent,
SipCore as well as for the entire system. Included in the statistics
are complete/incomplete calls, failed calls (based on user-defined
thresholds) and type of generated traffic. Call Statistics window
provide detailed call wise statistics per SipCore.
System statistics window provides the overall call statistics such as
active calls in progress, completed calls, number of successful
calls, attempted calls, and so on for each SipCore.
All events and statistics can be exported and saved for record or
review at a later time.
Figure: Statistics and Events
818 West Diamond Avenue - Third Floor, Gaithersburg, MD 20878, U.S.A
(Web) http://www.gl.com/ - (V) +1-301-670-4784 (F) +1-301-670-9187 - (E-Mail) [email protected]
Document Number: PKS100-14.7.25-01
Page 4
RTP Action Scripting
PacketGen™ provides a powerful scripting capability to control
RTP traffic. Scripting features includes loops, conditional
statements, wait for events, timers etc., Scripting gives the user
greater control over the RTP traffic being generated allowing
users to create/test IVR kind of applications. Scripts can be
created using the RTP Script Editor, which allows an intuitive,
point and click script setup.
The set of script elements included in the script editor allows user
to perform all the traffic generation / reception actions as done
using PacketGen™ main graphical user interface.
Voice Quality Testing using PacketGen™
PacketGen™ can be used to establish calls and send/record voice
files over the IP network. These voice files are then analyzed using
GL’s Voice Quality Testing (VQT) application as per ITU algorithms.
Voice Quality testing can be completely automated using
PacketGen™ CLI, RTP Action scripting, along with ASR Listener,
FCU and VQT software.
Command Line Interface
In addition to the GUI, PacketGen™ can also be operated through
a Command Line Interface (CLI). All the functionalities of the
PacketGen™ GUI are supported, except the configuration
functions. Users can thus operate PacketGen™ from a DOS based
console (instead of the GUI) or easily integrate PacketGen™ into
their own applications.
Figure: Command Line Interface
Figure: RTP Action Scripting
Audio File Converter Utility (AFC)
PacketGen™ now transmits and records voice files using a GL
proprietary file format (.glw). The accompanying GL Audio File
Converter Utility (AFC) will automatically convert any voice file,
into *.glw file format and vice versa. This allows the ability to
send/receive voice files at a higher density with multiple codecs
(the file is predefined with the desired codec). It also allows for
Discontinuous Transmission/Reception.
The Auto Audio FCU (part of AFCU) is generally used in
conjunction with GL's VQT application and converts degraded
voice files from their native codec format to a standard format
used by VQT. The Command line interface (CLI) in AFCU allows
the users to load, start, and stop Auto FCU configurations,
convert single file (raw / wav / glw file) of one codec to another
file format of a different codec using the commands.
Audio Stream Utility
The existing "Playback" feature is used to play the selected call to
speaker on the local computer (SIP/RTP core). To allow these calls
to be heard from remote systems, GL has introduced Audio
Stream Utility with PacketGen™. This utility automatically streams
the voice of a selected call to a speaker on a remote system.
Buyer’s Guide
PKG100 – PacketGen
PKS100 – PacketGen™ (includes PacketScan™)
PKS101 – SIP Core (additional)
PKS102 – RTP Soft Core (additional)
PCD103 – AMR Codec for PacketGen™
PCD104 – EVRC Codec for PacketGen™
PCD105 – EVRC-B Codec for PacketGen™
Related Software
PKV100 – PacketScan™ (Online and Offline)
PKB100 – RTP ToolBox™
PKS120 – MAPS™ SIP Emulator
PKS121 – SIP Conformance Test Suite (Test Scripts)
PKV107 – LTE Analyzer
PKV105 – SIGTRAN Analyzer
IPN400 – IPNetSim™ – 1Gbps w/ 4 links through bandwidth
FXT002 – GL Insight™ – Single Fax Analysis – IP
MDT002 – GL Insight™ – Single Modem Analysis – IP
PKS150 – TDM / VoIP Gateway w/ Analog and Digital Interfaces
VQT004 – Voice Quality Testing (PAMS, PSQM, PESQ)
VQT013 – VQuad™ with SIP (VoIP) Call Control
VBA032 – Near Real-time Voice-band Analyzer
*Specifications are subject to change without notice.
818 West Diamond Avenue - Third Floor, Gaithersburg, MD 20878, U.S.A
(Web) http://www.gl.com/ - (V) +1-301-670-4784 (F) +1-301-670-9187 - (E-Mail) [email protected]
Document Number: PKS100-14.7.25-01

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